Voice communications are increasingly shifting to web and Internet based applications that are outside of traditional telephony networks. Enterprise users desire to access their unified communications applications with their own Internet connected mobile devices, and consumers increasingly prefer Internet based communications channels for accessing contact centers.
Some communications service providers (“CSPs”) and enterprises have deployed real-time communications (“RTC”) applications based on a technology known as “WebRTC.” WebRTC is an open Internet standard for embedding real-time multimedia communications capabilities (e.g., voice calling, video chat, peer to peer (“P2P”) file sharing, etc.) into a web browser. For any device with a supported web browser, WebRTC can use application programming interfaces (“APIs”) to equip the device with RTC capabilities without requiring users to download plug-ins. By using WebRTC, CSPs may create new web based communications services and extend existing services to web based clients.
Currently, in order to be able to receive incoming calls, a WebRTC application of a user equipment (“UE”) maintains a continuous ongoing communications session. This may result in resource utilization issues for the UE and/or the network. For example, on the UE side, a continuous communications session utilizes resources such as battery and data of the UE, and on the server side, a continuous communications session utilizes resources such as network ports of the server to which the UE is connected.